Communication devices such as User Equipments (UE) are enabled to communicate wirelessly in a radio communications system, sometimes also referred to as a radio communications network, a mobile communication system, a wireless communications network, a wireless communication system, a cellular radio system or a cellular system. The communication may be performed e.g. between two user equipments, between a user equipment and a regular telephone and/or between a user equipment and a server via a Radio Access Network (RAN) and possibly one or more core networks, comprised within the wireless communications network.
User equipments are also known as e.g. mobile terminals, wireless terminals and/or mobile stations, mobile telephones, cellular telephones, or laptops with wireless capability, just to mention some examples. The user equipments in the present context may be, for example, portable, pocket-storable, hand-held, computer-comprised, or vehicle-mounted mobile devices, enabled to communicate voice and/or data, via the RAN, with another entity.
The wireless communications network covers a geographical area which is divided into cell areas, wherein each cell area being served by a network node such as a Base Station (BS), e.g. a Radio Base Station (RBS), which sometimes may be referred to as e.g. eNB, eNodeB, NodeB, B node, or BTS (Base Transceiver Station), depending on the technology and terminology used. The base stations may be of different classes such as e.g. macro eNodeB, home eNodeB or pico base station, based on transmission power and thereby also cell size. A cell is the geographical area where radio coverage is provided by the base station at a base station site. One base station, situated on the base station site, may serve one or several cells. Further, each base station may support one or several radio access and communication technologies. The base stations communicate over the radio interface operating on radio frequencies with the user equipments within range of the base stations.
In some RANs, several base stations may be connected, e.g. by landlines or microwave, to a radio network controller, e.g. a Radio Network Controller (RNC) in Universal Mobile Telecommunications System (UMTS), and/or to each other. The radio network controller, also sometimes termed a Base Station Controller (BSC) e.g. in GSM, may supervise and coordinate various activities of the plural base stations connected thereto. GSM is an abbreviation for Global System for Mobile Communications (originally: GroupeSpécial Mobile).
In the context of this disclosure, the expression Downlink (DL) is used for the transmission path from the base station to the user equipment. The expression Uplink (UL) is used for the transmission path in the opposite direction i.e. from the user equipment to the base station.
In 3rd Generation Partnership Project (3GPP) Long Term Evolution (LTE), base stations, which may be referred to as eNodeBs or even eNBs, may be directly connected to one or more core networks.
UMTS is a third generation mobile communication system, which evolved from the GSM, and is intended to provide improved mobile communication services based on Wideband Code Division Multiple Access (WCDMA) access technology. UMTS Terrestrial Radio Access Network (UTRAN) is essentially a radio access network using wideband code division multiple access for user equipments. The 3GPP has undertaken to evolve further the UTRAN and GSM based radio access network technologies.
According to 3GPP/GERAN, a user equipment has a multi-slot class, which determines the maximum transfer rate in the uplink and downlink direction. GERAN is an abbreviation for GSM EDGE Radio Access Network. EDGE is further an abbreviation for Enhanced Data rates for GSM Evolution.
Jitter buffers or de-jitter buffers are used to counter jitter introduced by queuing in packet switched networks so that a continuous play out of audio or video transmitted over the network is ensured. A jitter buffer delays the play out of each audio frame or video frame for a certain period of time, which introduces an additional delay for the end-to-end flow, but the jitter buffer compensates for delay variations in the transport.
If data packets are delayed long enough to allow the data packet with the highest transport delay to arrive before its scheduled play out time, the receiver can make a proper reconstruction of the signal. However, this will cause an unacceptable long mouth-to-ear delay for the call. On the other hand, if the jitter buffer is very short or non-existing the network delay variations will cause some data packets to arrive too late for their scheduled play out time, causing interruptions in the speech.
A jitter buffer can either be of a fixed size, or have the capability of adjusting its size dynamically in order to optimize the tradeoff between added delay and late losses. The expression “late losses” when used herein refers to data packets arriving too late for their scheduled play out time. A jitter buffer with the capability of adjusting its size dynamically, as mentioned above, is called an adaptive jitter buffer.
An adaptive jitter buffer continuously adjusts the jitter buffer depth according to measurements of jitter for previous packets in the flow.
FIG. 1 shows an illustration of an exemplary structure of a prior art MTSI speech receiver 100 comprising an adaptive jitter buffer 102 that receives audio or video payload such as Real-time Transport Protocol (RTP) payload. Reference to FIG. 1 will be made in order to clarify some terminology and the relation between different functional components of the receiver 100. The receiver 100 comprises further a jitter buffer adaptation logic 104 and a network analyzer 106. The jitter buffer adaptation logic 104 uses measurements of packet arrival times received from the network analyzer 106 and current jitter buffer status to determine if the buffer size needs to be adjusted, and as a result of that whether the play out time of some speech frames should be compressed or expanded. The receiver 100 comprises also an adaptation unit 108 and a speech decoder 110 which may perform compression or expansion of speech frames by performing so called time scaling, which may lead to impairments of the speech quality.
3GPP defines some minimum requirements that a jitter buffer algorithm in MTSI must fulfill. The exact algorithm is implementation dependent.
Handover in a Communication System
The concept of a communication system is that it has large number base stations covering a geographical area, providing mobility to the users. As a result it is a very basic requirement of the communications system that as the mobile handsets move from one cell to another, it must be possible to handover the call or the data session over from one base station to another with small disruptions.
The handover (HO) may occur between cells within the same Radio Access Technology (RAT), e.g., when a connection is handed over from an LTE cell to another LTE cell. Also an inter-RAT handover may happen, e.g., from an LTE cell to a WCDMA cell.
The purpose of the Intra-LTE handover feature is to manage the handover of a user equipment from one LTE cell to another LTE cell. This ensures that the user equipment is being served by the best cell at all times, and avoids call drops as the user equipment moves out of coverage of one LTE cell into the coverage of another LTE cell. The handover in LTE is network controlled based upon user equipment measurement reports of the serving and neighboring cells.
The intra-LTE handover process is managed by the radio base station. As previously mentioned, the radio base station is named eNodeB in LTE. The source eNodeB and target eNodeB communicate through an X2 link if there is one between the two eNodeBs. When an X2 link is not operable, the eNodeBs communicate over the Mobility Management Entity (MME) through S1 links.
FIG. 2 shows the procedure of an intra-LTE handover according to 3GPP TS 36.300. The handover procedure can be divided into three phases: the measurement phase, the preparation phase, and the execution phase.
Measurement Phase
In step 1 of FIG. 2, the network sends out measurement control, which sets different parameters related to handover to the user equipment (UE). One of the parameters is named handover hysteresis, or “HO hysteresis”, and another parameter is called “Time To Trigger” (TTT).
The UE makes periodic measurements of Radio Signal Received Power (RSRP) and Radio Signal Received Power (RSRQ) based on the radio signal received from the serving cell and from adjacent cells. In case the handover algorithm is based on RSRP values, when the RSRP value from an adjacent cell is higher than the one from the serving cell by a number of dBs equal to HO hysteresis, a timer is started and the UE triggers an Enters Event3A which means the UE is prepared to trigger a 3A event, which is the intra-frequency HO triggering event in LTE accordingly to 3GPP TS 36.300.
Once the timer is started, two things may happen:
A first thing that may happen during the TTT is that the RSRP of the serving cell becomes better than the best adjacent cell by a number of dBs equal to HO hysterisis. In this case the UE will stop the timer and triggers a Leave Event3A event so that the handover will not happen. The UE will continue to be served by the serving cell and nothing will be reported to the network.
A second thing that may happen is that TTT is reached without the serving cell becomes better than the best adjacent cell by a number of dBs equal to HO hysterisis. In step 2, the UE will send a measurement report to the network and request for handover.
The TTT and hysteresis are configuration parameters. The TTT is normally between 0 and 5120 ms according to 3GPP TS 36.331, with the default value set to 40 ms for Ericsson LTE networks according to L11B CPI. The hysteresis is in order of a few dBs with the default of 1 dB in Ericsson LTE networks according to L11B CPI.
Preparation Phase
The preparation phase is from when the eNodeB receives the measurement report in step 2, until the eNodeB sends out RRC connection reconfiguration in step 7. During this time, the source eNodeB prepares the handover by contacting target eNodeB. Also, the UE is reachable from the network during the phase.
From live measurements, we found that the average handover preparation time is 32 ms, which figure is based on more than 1 million samples of recent data.
Execution Phase
The execution phase contains an outage time where neither the source cell nor the target cell can reach the UE. That is during handover execution between step 7 and step 11 in FIG. 2. Data packets arriving at the source cell during the outage time are either dropped or forwarded to the target cell, where the target cell will send the data packets to UE when it is reconnected. The latter approach is taken by current Ericsson LTE networks, where packet forwarding is implemented through the X2 interface.
Despite the use of data packet forwarding, the outage time during a LTE handover is in average about 90 ms according to measurements. Although some of the handovers have a significantly larger outage time, above 200 ms.
A real-time service like VoIP typically includes a jitter buffer algorithm which works fine in normal situations. However, handover in mobile networks may have a significant impact on the jitter buffer. During a handover, packets can be delayed 90 ms or more because of the handover outage time. The user equipment may not have enough frames in the jitter buffer to deal with this, resulting in buffer under-run which gives bad media quality to the end-user.
US patent US 2006/0245394 describes a method to improve application layer service quality during a handover situation. OSI layer 2-originated indications of a handover being imminent, a link down for handover, and a new link up after handover is signaled to an application operating at an OSI layer above layer 3. The indications may be used by the application to take proactive measures prior to the actual handover, e.g., buffer more data that can be used during the handover outage e.g. in the case of a non-real-time streaming application.
However, the time available from when the handover decision is taken until the handover actually happens may be quite small, from our investigation in LTE around 30 ms. For a real-time service like VoIP it is not possible to pre-buffer data to cover the handover outage time during this limited time interval.
There are a number of papers published on handover prediction, e.g. “A History-Based Handover Prediction for LTE Systems”, Huaining Ge; Xiangming Wen; Wei Zheng; Zhaoming Lu; Bo Wang; International Symposium on Computer Network and Multimedia Technology, 2009, and“Comparison of User Mobility Pattern Prediction Algorithms to increase Handover Trigger Accuracy”, Michaelis, S. Wietfeld, C, Vehicular Technology Conference, 2006. In these papers UE movement prediction is used as an additional information compared to classical handover preparation and triggers. Some use statistical analysis, others complex pattern detection algorithms or historic analysis to predict future movements of the UEs. The target of the proposed handover prediction is to aid the triggering of cell handovers, thereby improving the handover performance from a network point of view, for example by minimizing the number of ping-pong handovers in the communications system. By ping-pong handovers is herein meant several handovers between the same two radio base stations within a certain period of time. The certain period of time may vary between different radio technologies.
Using handover prediction algorithms to improve handover performance in the communications system may be very useful, for example to remove unnecessary ping-pong handovers, thereby increasing the overall service quality of the users. However, handovers are a normal and necessary part of a mobile communications system, and even if all necessary cases are removed by smart algorithms, real-time applications may still suffer performance problems due to necessary handovers in the communications system.